INTERNET-DRAFT L. Coene(Ed)
Internet Engineering Task Force Siemens
Issued: March 2003 J. Pastor
Expires: September 2003 Ericsson
Telephony Signalling Transport over SCTP applicability statement
<draft-ietf-sigtran-signalling-over-sctp-applic-08.txt>
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Abstract
This document describes the applicability of the new protocols
developed under the signalling transport framework[RFC2719]. A
description of the main issues regarding the use of the Stream
Control Transmission Protocol (SCTP)[RFC2960] and each adaptation
layer for transport of telephony signalling information over IP
infrastructure is explained.
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Draft Telephony Signalling over SCTP AS March 20031 INTRODUCTION
This document intends to inform how to transport telephony
signalling protocols, used in classic telephony systems, over IP
networks. The whole architecture is called SIGTRAN (Signalling
Transport) as described in RFC2719 and is composed of a transport
protocol(SCTP) and several User Adaptation layers(UAL). The
transport protocol SCTP has been been developed to fulfill the
stringent requirements that telephony signalling networks have. The
set of User Adaptation layers have also been introduced to make it
possible that different signalling protocols can use the SCTP layer.
1.1 Scope
The scope of this document is to explain the way that user
adaptation layers and SCTP protocols have to be used to transport
Telephony signalling information over IP.
1.2 Terminology
The following terms are commonly identified in related work:
Association: SCTP connection between two endpoints.
Stream: A uni-directional logical channel established within an
association, within which all user messages are delivered in
sequence except for those submitted to the unordered delivery
service.
SPU: Signalling protocol user, the application on top of the User
adaptation layer.
CTSP: Classical Telephony Signalling protocol(examples: MTP level2,
MTP level 3, SCCP....).
UAL: User adaptation layer: the protocol that encapsulate the upper
layer telephony signalling protocols that are to be transported over
SCTP/IP.
ISEP: IP signalling endpoint: a IP node that implements SCTP and a
User adapatation layer.
SP: signalling point
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Draft Telephony Signalling over SCTP AS March 2003
- SS7 SCCP users: RANAP, MAP(+TCAP), INAP(+TCAP)...
- ISDN Q.921 users: Q.931
- V5.2/DSS1
- ....
Every classic telephony protocol can have a corresponding UAL
developed.
The user adaptation layers(UALs) are a set of protocols that
encapsulate a specific signalling protocol to be transported over
SCTP. The adapation is done in a way that the upper signalling
protocols that are relayed remain unaware that the lower layers are
different to the originail lower telephony signalling layers. In
that sense, the upper interface of the user adapatation layers need
to be the same as the upper layer interface to its original lower
layer. If a MTP user is being relayed over the IP network, the
related UAL used to transport the MTP user will have the same upper
interface as MTP has.
The Stream Control Transmission protocol was designed to fulfill the
stringent transport requirements that classical signalling protocols
have and is therefore the recommended transport protocol to use for
this purpose.
The following functions are provided by SCTP:
- Reliable Data Transfer
- Multiple streams to help avoid head-of-line blocking
- Ordered and unordered data delivery on a per-stream basis
- Bundling and fragmentation of user data
- Congestion and flow control
- Support continuous monitoring of reachability
- Graceful termination of association
- Support of multi-homing for added reliability
- Protection against blind denial-of-service attacks
- Protection against blind masquerade attacks
SCTP is used as the transport protocol for telephony signalling
applications. Message boundaries are preserved during data
transport by SCTP and so each UAL can specify its own message
structure within the SCTP user data. The SCTP user data can be
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Draft Telephony Signalling over SCTP AS March 2003
delivered by the order of transmission within a stream(in sequence
delivery) or unordered.
SCTP can be used to provide redundancy at the
transport layer and below. Telephony applications needing this level
of redundancy can make use of SCTP's multi-homing support.
SCTP can be used for telephony applications where head-of-line
blocking is a concern. Such an application should use multiple
streams to provide independent ordering of telephony signalling
messages.
3 Issues for transporting telephony signalling over SCTP
Transport of telephony signalling requires special
considerations. In order to use SCTP, special care must be taken to
meet the performance, timing and failure management requirements.
3.1 Congestion Control
The basic mechanism of congestion control in SCTP have been
described in [RFC2960]. SCTP congestion control sometimes conflicts
with the timing requirements of telephony signalling application
messages which are transported by SCTP. During congestion, messages
may be delayed by SCTP, thus sometimes violating the timing
requirements of those telephony applications.
In an engineered network (e.g. a private intranet), in which network
capacity and maximum traffic are very well understood, some
telephony signalling applications may choose to relax the congestion
control rules of SCTP in order to satisfy the timing
requirements. In order to do this, they should employ their own
congestion control mechanisms. But this should be done without
destabilising the network, otherwise this would lead to potential
congestion collapse of the network.
Some telephony signalling applications may have their own congestion
control and flow control techniques. These techniques may interact
with the congestion control procedures in SCTP.
3.2 Detection of failures
Telephony systems often must have no single point of failure in
operation.
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Draft Telephony Signalling over SCTP AS March 2003
The UAL must meet certain service availability and performance
requirements according to the classical signalling layers they are
replacing. Those requirements may be specific for each UAL.
For example, telephony systems are often required to be able to
preserve stable calls during a component failure. Therefore error
situations at the transport layer and below must be detected quickly
so that the UAL can take approriate steps to recover and preserve the
calls. This poses special requirements on SCTP to discover
unreachablility of a destination address or a peer.
3.2.1 Retransmission TimeOut (RTO) calculation
The SCTP protocol parameter RTO.Min value has a direct impact on the
calculation of the RTO itself. Some telephony applications want to
lower the value of the RTO.Min to less than 1 second. This would
allow the message sender to reach the maximum
number-of-retransmission threshold faster in the case of network
failures. However, lowering RTO.Min may have a negative impact on
network behaviour [ALLMAN99].
In some rare cases, telephony applications might not want to use the
exponential timer back-off concept in RTO calculation in order to
speed up failure detection. The danger of doing this is that, when
network congestion occurs, not backing off the timer may worsen the
congestion situation. Therefore, this strategy should never be used
in public Internet.
It should be noted that not using delayed SACK will also help faster
failure detection.
3.2.2 Heartbeat
For faster detection of (un)availability of idle paths, the
telephony application may consider lowering the SCTP parameter
HB.interval. It should be noted this might result in a higher traffic
load.
3.2.3 Maximum number of retransmissions
Setting Path.Max.Retrans and Association.Max.Retrans SCTP parameters
to lower values will speed up both destination address and peer
failure detection. However, if these values are set too low, the
probability of false fault detections might increase.
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Draft Telephony Signalling over SCTP AS March 20033.3 Shorten end-to-end message delay
Telephony applications often require short end-to-end message
delays. The method described in section 3.2.1 on lowering RTO may
be considered. The different paths within a single association will
have a different RTO, so using the path with the lowest RTO will
lead to a shorter end-to-end message delay for the application
running on top of the UAL's.
3.4 Bundling considerations
Bundling small telephony signalling messages at transmission helps
improve the bandwidth usage efficiency of the network. On the
downside, bundling may introduce additional delay to some of the
messages. This should be taken into consideration when end-to-end
delay is a concern.
3.5 Stream Usage
Telephony signalling traffic is often composed of multiple,
independent message sequences. It is highly desirable to transfer
those independent message sequences in separate SCTP streams. This
reduces the probability of head-of-line blocking in which the
retransmission of a lost message affects the delivery of other
messages not belonging to the same message sequence.
4. User Adaptation Layers
Users Adaptation Layers (UALs) are defined to encapsulate different
signalling protocols in order to transport them over SCTP/IP
There are UALs for both access signalling (DSS1) and trunk signalling
(SS7). A brief description of the standardized UALs follows in the
next sub-sections.
The delivery mechanism in the several UALs
- Supports seamless operation of UALs user peers over an IP
network connection.
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Draft Telephony Signalling over SCTP AS March 2003
- Supports the interface boundary that the UAL user had with the
traditional lower layer.
- Supports management of SCTP transport associations and traffic
between SGs and ISEPs or two ISEPs
- Supports asynchronous reporting of status changes to management.
Signalling User Adaptation Layers have been developed for both:
Access and Trunk Telephony Signalling. They are defined as follows.
Access Signalling: This is the signalling that is needed between and
access device and an exchange in the core network in order to
establish, manage or release the voice or data call paths. There
are several protocols that have been developed for this purpose.
Trunk Signalling: This is the signalling that is used between the
exchanges inside the core network in order to establish, manage or
release the voice or data call paths. The most common protocols
used for this purpose are known as the SS7 system that belongs to
the Common Channel Signalling (CCS) philosophy. The SS7 protocol
stack is depicted below:
+------+-----+-------+- -+-------+------+-----+------+
| | | | | | MAP | CAP | INAP |
+ | + RANAP |...| BSSAP +-------------------+
| ISUP | TUP | | | | TCAP |
+ | +---------------------------------------+
| | | SCCP |
+----------------------------------------------------+
| MTP3 |
+----------------------------------------------------+
| MTP2 |
+----------------------------------------------------+
| MTP1 |
+----------------------------------------------------+
The Telephony Signalling Protocols to be transported with the already
designed UALS are:
- ISDN Q.921 Users: Q.931
- V5.2/DSS1
- DPNSS/DASS2
- SS7 MTP3 Users: SCCP, ISUP, TUP
- SS7 MTP2 Users: MTP3
- SS7 SCCP Users: TCAP, RANAP, BSSAP, ...
Two main scenarios have been developed to use the different UALS for
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Draft Telephony Signalling over SCTP AS March 2003
This is also referred to as IPSP communication. IPSP stands for IP
Signalling Point and describes the role that the UAL plays on a
IP-based node.
The first scenario is applied for both types of signalling (access
and trunk signalling). On the other hand the peer to peer basis can
only be used for trunk signalling.
4.1 Access Signalling
The SIGTRAN WG have developed UALs to transport the following Access
Signalling protocols:
- ISDN Q.931
- V5.2
- DPNSS/DASS2
4.1.1 ISDN Q.931 over IP
UAL: IUA (ISDN Q.921 User Adaptation)
This document supports both ISDN Primary Rate Access (PRA) as well as
Basic Rate Access (BRA) including the support for both point-to-point
and point-to-multipoint modes of communication. This support
includes Facility Associated Signalling (FAS), Non-Facility
Associated Signalling (NFAS) and NFAS with backup D channel.
It implements the client/server architecture. The default orientation
is for the SG to take on the role of server while the ISEP is
the client. The SCTP (and UDP/TCP) Registered User Port Number
Assignment for IUA is 9900.
Examples of the upper layers to be transported are Q.931 and QSIG.
The main scenario supported by this UAL is the SG to ISEP
communication where the ISEP role is typically played by a node
called an MGC, as defined in [RFC2719].
****** ISDN ****** IP *******
*PBX *---------------* SG *--------------* MGC *
****** ****** *******
+-----+ +-----+
|Q.931| (NIF) |Q.931|
+-----+ +----------+ +-----+
| | | | IUA| | IUA |
| | | +----+ +-----+
|Q.921| |Q.921|SCTP| |SCTP |
| | | +----+ +-----+
| | | | IP | | IP |
+-----+ +-----+----+ +-----+
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Draft Telephony Signalling over SCTP AS March 20035 Security considerations
UALs are designated to carry signalling messages for telephony
services. As such, UALs must involve the security needs of several
parties: the end users of the services; the network providers and
the applications involved. Additional requirements may come from
local regulation. While having some overlapping security needs, any
security solution should fulfill all of the different parties'
needs. See specific Security considerations in each UAL technical
specification.
SCTP only tries to increase the availability of a network. SCTP does
not contain any protocol mechanisms which are directly related to
communication security, i.e. user message authentication, integrity
or confidentiality functions. For such features, it depends on
security protocols. In the field of system security, SCTP includes
mechanisms for reducing the risk of blind denial-of-service attacks
as it is described in section 11 in RFC2960.
This document does not add any new components to the protocols
included in the discussion. For secure use of the SIGTRAN protocols
the readers should go through the "Security Considerations for
SIGTRAN protocols" [RFCSIGSEC]). According to that document, the use
of the IPsec is the main recommendation to secure SIGTRAN protocols
in the Internet, but TLS is also considered as a perfectly valid
option to be used in certain scenarios. Recomendations of usage are
also included.
6 References and related work
[RFC2960] Stewart, R. R., Xie, Q., Morneault, K., Sharp, C. , ,
Schwarzbauer, H. J., Taylor, T., Rytina, I., Kalla, M., Zhang,
L. and Paxson, V, "Stream Control Transmission Protocol", RFC2960,
October 2000.
[RF3257] Coene, L., Tuexen, M., Verwimp, G., Loughney, J., Stewart,
R. R., Xie, Q., Holdrege, M., Belinchon, M.C., and Jungmayer, A.,
"Stream Control Transmission Protocol Applicability statement",
RFC3257, April 2002.
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Draft Telephony Signalling over SCTP AS March 2003
Lode Coene, John Loughney, Michel Tuexen, Randall R. Stewart,
Qiaobing Xie, Matt Holdrege, Maria-Carmen Belinchon, Andreas
Jungmaier, Gery Verwimp and Lyndon Ong.
The authors wish to thank Renee Revis, H.J. Schwarzbauer, T. Taylor,
G. Sidebottom, K. Morneault, T. George, M. Stillman, B. Bidulock
and many others for their invaluable comments.
8 Author's Address
Lode Coene Phone: +32-14-252081
Siemens Atea EMail: lode.coene@siemens.com
Atealaan 34
B-2200 Herentals
Belgium
Javier Pastor-Balbas Phone: +34-91-3393819
Ericsson Espana S.A. Email: j.javier.pastor@ericsson.com-
C/ Retama 1
28045 Madrid
Spain
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Draft Telephony Signalling over SCTP AS March 2003
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